1. Field of the Invention
The present invention relates generally to telecommunications networks, and particularly to monitoring the quality of packet voice transmissions in real-time.
2. Technical Background
New telecommunications technologies are emerging that employ packet switching instead of the traditional circuit switched technologies provided by the public switched telephone networks (PSTN). These packet switched technologies are being used to support the transmission of digitized voice signals over a data network such as the Internet. Providing telephone-like full duplex voice over an Internet Protocol network (VoIP) is particularly important. VoIP services are attractive to commercial long-distance carriers because they enable the use of global Internet transport facilities to carry traffic that is presently being carried over dedicated circuit switched facilities. The potential benefits of VoIP are enormous in terms of better utilization of network bandwidth to support telephone traffic and the economies of scale from the use of one kind of transport for all telecommunications services. However, a major impediment to the immediate adoption of VoIP services relates to the user perception of the quality of voice communications using VoIP.
Traditional telephone connections have been subject to impairments in the form of noise, attenuation, distortion, cross-talk and echo. Such impairments are particularly common in analog portions of the network, such as subscriber loops and frequency domain multiplexing equipment. Digital transmission alleviates many of these problems but also introduces quantization noise and distortions due to bit errors in the digital signal. Even with perfect digital transmission applied to long haul transmissions, a typical telephone connection still includes many analog components where impairments can occur.
A poor connection or malfunctioning piece of equipment can produce conditions that a telephone customer will find objectionable or intolerable. When there is a high incidence of poor connections, customers may complain to the service provider or to a regulatory authority, or simply change long distance carriers. Thus, the perceived quality of a service provider's service is a major factor affecting the reputation and marketability of long distance telephone services.
To guard against poor quality, service providers have developed methods to obtain objective quality measurements upon a line, piece of equipment, or an end-to-end telephone connection. These measurements can help the service provider detect and gauge impairments, pinpoint weak elements, and correct deficiencies that degrade user perception of quality. The effects of extreme fault conditions on user perception of quality is clear. There are easily discernable thresholds for “no effect” and “substantial degradation” conditions. As a result, the average consumer has to come to expect a certain quality of service from the PSTN.
With the proliferation of voice-over-packet technologies, maintaining a quality of service comparable to the PSTN is a major concern of service providers, equipment vendors, and ultimately the consumers of packetized telecommunications services. Unlike circuit switched traffic, real time voice transmission using packet switched technologies is sensitive to packet loss, packet delay, and packet jitter occurrences which are characteristic of packet switched networks. Packet loss and packet delay variations may impact the ability of a voice codec to faithfully reproduce a digitally encoded voice signal. When a received packetized voice transmission is missing some packets, a codec may provide an audio signal that is distorted, garbled, or otherwise degraded.
In one approach that has been considered, the IntServ and Diffserv protocols have been proposed for improving the reliability and consistency of packet transport. (For reference, the IntServ and Diffserv approaches are described in documents, RFC 1638 and 3317 respectively, promulgated by the Internet Engineering Task Force (IETF).)
The Real Time Control Protocol (RTCP) has also been considered for obtaining real-time measurements of the receipt of packet data, and for reporting the measurements to a sender or to a network quality monitoring location. (For reference, RTCP is described in IETF document RFC 1889 and in ITU Recommendation H.225.0.)
One drawback to these various approaches is that, while the reporting of packet arrival statistics provides some estimate of data transmission quality, there is no consideration of the extent to which packet loss, packet delay, and packet jitter affect the perceived quality of a reconstructed voice signal.
Another drawback relates to the fact that the manner in which data transmission quality affects perceived quality of a voice channel is often dependent on the coding scheme employed by the codec. Various codec schemes may exhibit differing susceptibilities to packet-transmission variations and the dependency may be quite non-linear for some codec schemes. For example, a given packet loss rate or jitter may have very little effect on a G.711 encoded speech signal, whereas an identical packet loss rate or jitter may seriously degrade a more sophisticated G.723 encoded signal. Even if the particular codec scheme is known, an accurate model of codec behavior is required to map the observed packet characteristics to a perceived quality level. The introduction of a new standard encoding scheme, or a proprietary encoding scheme or encryption scheme, would necessitate the development and deployment of new models.
For these reasons, the mere reporting of packet loss statistics is inadequate. Without accurate information about the perceived quality level being experienced by users, a service provider is not certain when corrective action is necessary or what corrective action needs to be taken to improve packet transmission performance. A service provider may shift traffic or resources unnecessarily to improve quality of service (QOS) when, in fact, the perceived level of degradation is negligible to the users of the service. Likewise, a seemingly minor packet delivery problem might be causing big problems in user perception due to the particular codec scheme being used.
What is needed is a means for automatically monitoring and reporting in near real-time the quality of a packet voice transmission as perceived by a user receiving voice communications.